om a prepaid account based on the last instant the
> mediaproxy saw an RTP packet.
>
> But why to force using mediaproxy with no choice? And why to force it for
> every call, whether it falls under CallControl's control or not?
>
> I am using Kamailio 3.2.
>
>
>
Hi,
I can see you posting multiple times on both proxies listings so I'm sure
you havent heard back from anyone.I am not at all familiar with your
functions in email but could it be possible for you to determine on which
calls you need to engage mediaproxy and on which not to, then on the base
of t
which version are you using, there is no such condition in this page or is
it?
http://kamailio.org/docs/modules/3.1.x/modules_k/dispatcher.html#id2821010
On Thu, Feb 16, 2012 at 2:31 PM, Mino Haluz wrote:
> Hi,
>
> is there any way how to reload dispatcher destinations (located in db)
> without
Can you send the output of following commands:
1# ifconfig
2# netstat -pln
3# route -n
I just want to see how is your VPN interface routing done and if a tunnel
interface is created your kamailio is listening on that interface or not !
Regards,
Sammy
On Thu, Feb 16, 2012 at 2:10 PM, 393319152 <
SIP: 400 Bad Request
2012/2/3 Albert Petit
> Hi,
>
>
> SIP UAS1
>
>
> SIP LB <->
>
>
>
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users maili
Hi,
Kamailio is definitely the exact tool for this purpose, I have exactly the
same setup running as yours and for scalability we started using Kamailio
in front of our asterisk servers. Long story short, read these articles.
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-as
erface eth0:0 and
same for Public IP.
That was all, everything working great. I could use the physical IP as well
as the floating IP both on same setup with exception of RTPproxy not
picking up the physical IPs due to the default route thing.
Regards,
Sammy
On Fri, Jan 27, 2012 at 2:36 PM, Sammy
Hi,
Can you copy/paste the error message here. I've recently made kamailio work
with floating IP(Heartbeat) and I was having similar issue but then I
resolved due to help from community memebrs here.
This is how I made kamailio work with floating IP.
in */etc/sysctl.conf* file insert this line
*ne
also for doing upgrades in the
> configuration in a controlled manner.
> Hope it helps
>
> Regards
>
> Javi
>
>> --
>>
>> Message: 6
>> Date: Thu, 26 Jan 2012 12:13:18 +0500
>> From: Sammy Govind
>> Subject: Re: [SR-U
49 PM, Andreas Granig wrote:
> Hi Sammy,
>
> On 01/25/2012 02:08 PM, Sammy Govind wrote:
> > I've been looking for ways to create a redundant Kamailio cluster. I've
> > googled alot but haven't got any concrete or final wording on any one
> > solution that&#
Hello list,
I've been looking for ways to create a redundant Kamailio cluster. I've
googled alot but haven't got any concrete or final wording on any one
solution that'll just work perfectly. The basic requirement is that in case
of Kamailio application failure or in case of physical machine error
> tried to allow all packets going in/out of these interfaces but no luck. It
> seems its routing problem.
>
> *From:* Sammy Govind
> *To:* nunu abe
> *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List
> *Sent:* Tuesday, January
e expanding the wireshark message, I see "Unrecognised SIP
> header".**
> User-Agent: snom370/7.3-boco-test
> Accept: application/sdp
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,
> MESSAGE, INFO
> Allow-Events: talk, hold, refer,
header.
> Any ideas ???
>
> Thank you for your help?
>
> Regards
> Maedot
> *From:* Sammy Govind
> *To:* nunu abe
> *Cc:* SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
> Users Mailing List
> *Sent:* Friday, January 6, 2012 12:02 PM
>
> *Su
o/kamailio.cfg]
> l=796 a=40 n=setflag
> 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg]
> l=799 a=3 n=return
> 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg]
> l=470 a=6 n=route
> 4(2458) ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamail
Don't forget to change the modparams for RTPproxy as
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722")
# modparam("rtpproxy", "rtpproxy_sock",
"unix:/var/run/rtpproxy/rtpproxy.sock")
On Fri, Jan 6, 2012 at 2:05 PM, Sam
an 6, 2012 at 1:31 PM, nunu abe wrote:
> Hi,
>
> I have already posted this message but it was sent only to Sammy Govind
> somehow, not to the mailing list. So I am posting it again to the list. I
> am sorry if this is double posting or if this is not the right way to do i
pts/1S+ 15:31 0:00 grep
> --color=auto rtpproxy
>
> x@DualStackCS:~$ sudo /etc/init.d/kamailio start
>
>
> 6. Started kamailio
> @DualStackCS:~$ sudo /etc/init.d/kamailio restart
>
> I have attached the log file.
>
> Many thanks for your help.
>
>
Hi,
I had a good laugh when I read the reply, but going through the whole
thread again I found this:
" It displays the same message even when I replace ADDR_IPV4 and ADDR_IPV6
with IP addresses. "
So there _IS_ some problem here.
Nuno, can you please copy paste the exact command which you execut
t times yes and times
> now so I am not sure it is a definite DB issue, the servers are in the
> same LAN, and there are other modules that use the DB string that work
> when launched before this one while this one does not work.
> Regards
>
> On Sun, Jan 1, 2012 at 4:13 PM
Hi,
forget the whole list of errors and just resolve the DB connectivity
between the Kamailio server and DB server first .
27(24762) ERROR: db_mysql [km_my_con.c:109]: driver error: Can't
> connect to MySQL server on 'x' (4)
try connecting to the remote DB server 'x' from linux s
Yes
On Tue, Dec 20, 2011 at 10:50 PM, Ahmed Yousef wrote:
>
>
> is KAMAILIO work as registrar server
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-
rver)), then maybe you just
> use dialog module with dlg_bridge command:
>
> http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2554844
>
> Cheers,
> Daniel
>
>
> On 12/16/11 1:35 PM, Sammy Govind wrote:
>
> Hi,
> Thanks alot, I found the same stu
8 0\r\na=rtpmap:8
> PCMA/8000\r\na=rtpmap:0 PCMU/8000\r\n"`
> "
>
> EOF
>
> Cheers,
> Daniel
>
>
>
> On 12/16/11 10:02 AM, Sammy Govind wrote:
>
> Hello list,
>
> I'm using Kamailio 3.2 and am trying to send a SIP request from fifo t
Hello list,
I'm using Kamailio 3.2 and am trying to send a SIP request from fifo to
kamailio using below format.
#kamctl fifo t_uac_dlg MESSAGE sip:103@ mydomain.org . . "From:
1...@mydomain.org To: 1...@mydomain.org Contact: mydomain.org Content-Type:
text/plain; charset=UTF-8" "INVITE"
But I
Hi,
As per my practical experience the same file works for 3.2.z as I was using
for 3.1.x, nothing changed in configuration file except the part where
rtpproxy functions are called.
Regards,
Sammy.
On Thu, Dec 15, 2011 at 9:12 AM, Anthony Sanchez wrote:
> Is there an update of default Kamailio
wrote:
> Hello,
>
>
> On 12/2/11 5:24 AM, Sammy Govind wrote:
>
> Hello again,
>
> You were right, as soon as I made changes in asterisk SIP profile for
> the Kamailio proxy server and stopped the 401 Auth from Asterisk to
> Kamailio the CANCELS started to work fine.
>
&g
- ack
- invite with credentials, kamailio forwards to asterisk
- asterisk starts processing the invite and call can be cancelled now.
Thanks alot
--
Best Regards,
Sammy.
On Thu, Dec 1, 2011 at 12:01 PM, Sammy Govind wrote:
> Hey Daniel,
>
> I've exactly followed your point, I'l
ep traces since many times
> I am not around a computer where is convenient to open pcap file. On the
> other hand, if it is a transmission problem (at transport layer), pcap file
> is better.
>
> Cheers,
> Daniel
>
> On 11/29/11 5:07 AM, Sammy Govind wrote:
>
> Hell
Hi,
whats the output for the command "netstat -pln | grep 7722" or "netstat
-pln | grep -i rtpproxy". Just curious to know if rtpproxy is actually
listening on the mentioned port on localhost !
Regards,
Sammy
On Tue, Nov 29, 2011 at 2:37 AM, pablo umanzor wrote:
> which version are you using
calls are routed to MSs and
then comeback for further dial-outs.
Please see the Continuous CANCEL requests which aren't terminating the call.
Thanks,
Sammy.
On Mon, Nov 28, 2011 at 4:41 PM, Sammy Govind wrote:
> Thanks for your reply I will attach the wireshark traces as soon as I get
&
-W byline port 5060
>
> The sip trace will help to see what is wrong with that CANCEL.
>
> Cheers,
> Daniel
>
>
> On 11/28/11 7:19 AM, Sammy Govind wrote:
>
> Anyone please help.
>
> On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind wrote:
>
>> Hello list,
Anyone please help.
On Sat, Nov 26, 2011 at 10:39 PM, Sammy Govind wrote:
> Hello list,
>
> I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio handles
> all the SIP registrations. Calls from SIP phones are forwarded to asterisks
> and then dialled out t
Hello list,
I'm using Kamailio 3.1.5 in front of asterisk servers. Kamailio handles all
the SIP registrations. Calls from SIP phones are forwarded to asterisks and
then dialled out to Kamailio.
root@SBCserver:~# kamailio -V
version: kamailio 3.1.5 (x86_64/linux) 76fff5
flags: STATS: Off, USE_IPV6
Just using your configuration file for minimal changes
1- Add this line in the defines
#!define WITH_PSTN
2- Change the lines where PSTN _ GW IP/PORT are defined
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
pstn.gw_ip = "10.0.0.101" desc "
You can take sip trace using sipgrep/ngrep or tcpdump/tshark. Also I was
thinking if its just a matter of port forwarding then why deploy Kamailio
why not just use two/three iptables lines and forward to asterisk IP:PORT
!!?
Hhow about you follow some asterisk-kamailio integration post and change
Hi,
Can you capture a sip trace on Kamailio server for this, also paste your
configuration file ! so someone can figure out.
Regards,
Sammy.
On Tue, Nov 22, 2011 at 5:14 PM, arif tuhin wrote:
> I'm Trying to relay sip messages from sip client to asterisk. Kamailio
> should accept the msg on 90
until I finally make it work.
Any comments/suggestions from the List users is highly appreciated.
Regards,
Sammy
On Fri, Nov 4, 2011 at 7:08 PM, Klaus Darilion wrote:
> On 04.11.2011 10:57, Sammy Govind wrote:
>
>> SORRY for PM but this is not getting through to the users Li
Thanks Alex for actually finding the thread for me- I'm reading these
thoroughly and let me try these.
Klaus, mhome=1 is a good tip let me try it now: The problem is that when
I set-up the scenario as I mentioned using Private IPs the end user starts
sending the media to the private IP of asterisk
d perhaps you can
> just have your asterisk on a public IP. As the extra security you might
> think you have because it has a private IP, is much less then what the
> diagram says.
>
> Anyhow all is possible, the beauty of flexibility :)
>
>
>
> 2011/11/4 Sammy Gov
Hello List,
I've successfully configured Kamailio with a farm of asterisk servers in
L/B | F/O mode. The topology is something like below:
User<-->Kamailio<-->Asterisk-Servers
Both kamailio and Asterisk serevrs are on public IPs
*The problem:*
Kamailio and Asterisk Servers need to be on Public I
Thanks Alex,
Hey Uri, can you check if you've /etc/init.d/kamailio file exists - open
that file if exists and cross verify few parameters.
Few of these are as follows
PATH=/sbin:/bin:/usr/sbin:/usr/bin
DAEMON=*/usr/local/sbin/kamailio*
NAME=kamailio
DESC=kamailio
HOMEDIR=/var/run/
*PIDFILE*=$HOME
if chkconfig isn't working you can add the kamalio startup command in
etc/rc.local file.
On Mon, Oct 24, 2011 at 6:03 PM, Uri Shacked wrote:
> Hi,
> what is the best way to start kamailio at boot?
> the example in the install is not working for me
>
>
>
>
You may need to setup kamctlrc script in /usr/local/etc/kamailio/ for this
FIFO error.
On Sat, Oct 15, 2011 at 4:07 PM, Peter Schrock wrote:
> I tried opening with "kamctl moni" and I get this error message:
> ERROR: Error opening Kamailio's FIFO FIFO
> ERROR: Make sure you have the line 'modpara
Sorry for jumping in, it seems to me that its Domain name issue. are you
sure sip.my-domain.com resolves to your Kamailio Server. Is this domain
added in domain table and in SIP_DOMAIN env variable !!?
2011/10/10 Henrik Aagaard Sørensen
> When trying to dial 101 this is a tshark output on the Ka
0
> ;rport=5060;branch=z9hG4bK091005111656091709252938
> From: sip:919731573290@134.121.32.130;tag=09100511163117092280006157
> Call-ID: b637fa62393a45a0a58633c1a8f43a86
> To: sip:austin@134.121.32.130;tag=8c2e350c064e417c96bda1378470fd46
> Content-Length: 0
>
>
> SIP/2.0 2
Hey,
Can you send in the SIP/SDP invites. I suspect the codecs issue here.
--
Regards,
Sammy
On Sun, Oct 9, 2011 at 8:57 AM, Austin Einter wrote:
> Hi
> I am using Kamailio 3.1.5 . I am using RTP proxy also.
> I have used default kamailio.cfg.sample fiile , and just added line
> #!define WITH_NA
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